This is a rework of the audio join procedure whithout the explict listen
only separation in mind. It's supposed to be used in conjunction with
the transparent listen only feature so that the distinction between
modes is seamless with minimal server-side impact. An abridged list of
changes:
- Let the user pick no input device when joining microphone while
allowing them to set an input device on the fly later on
- Give the user the option to join audio with no input device whenever
we fail to obtain input devices, with the option to try re-enabling
them on the fly later on
- Add the option to open the audio settings modal (echo test et al)
via the in-call device selection chevron
- Rework the SFU audio bridge and its services to support
adding/removing tracks on the fly without renegotiation
- Rework the SFU audio bridge and its services to support a new peer
role called "passive-sendrecv". That role is used by dupled peers
that have no active input source on start, but might have one later
on.
- Remove stale PermissionsOverlay component from the audio modal
- Rework how permission errors are detected using the Permissions API
- Rework the local echo test so that it uses a separate media tag
rather than the remote
- Add new, separate dialplans that mute/hold FreeSWITCH channels on
hold based on UA strings. This is orchestrated server-side via
webrtc-sfu and akka-apps. The basic difference here is that channels
now join in their desired state rather than waiting for client side
observers to sync the state up. It also mitigates transparent listen
only performance edge cases on multiple audio channels joining at
the same time.
The old, decoupled listen only mode is still present in code while we
validate this new approach. To test this, transparentListenOnly
must be enabled and listen only mode must be disable on audio join so
that the user skips straight through microphone join.
Listen only has a built-in retry routine on join failures that's
convoluted half-broken. It stems from the Kurento era where it could
fail randomly due to a myriad of reasons.
Production logs indicate that the retry is seldom used nowadays in
mediasoup-based environments. The presence of the retry also breaks
the error troubleshooting modal when actual failures happening, leaving
users in the dark about what's happening.
Remove the listen only retry code from AudioManager and bubble up any
join failure to the callers.
- Adds a new Help view for unknown error codes
- Correctly detect NotAllowedError (permissions) - they are currently
being treated like unknown errors in the Help modal
- Rephrase NotAllowedError help text; make it more succint and direct
- Rephrase the unknown error help text; make it more succint and direct
- Add error code and message to that view
- Add public.media.audioTroubleshootingLinks to allow referencing KB
links on the Help modal
- See inline docs
public.media.showVolumeMeterInSettings => public.media.showVolumeMeter
public.media.simplifiedEchoTest => public.media.localEchoTest.enabled
Initial hearing state can be configured in public.media.localEchoTest.initialHearingState
New features:
- A simplified echo test mode that only does a local loopback (instead of
going to FS and back)
- A volume meter for microphone streams to the AudioSettings view
Those two features are experimental and disabled by default; see
public.app.media.simplifiedEchoTest and public.app.media.showVolumeMeter configs
Collateral changes:
- fix: localize fallback device strings in AudioSettings/DeviceSelector
- Refactor on some media stream utils to be re-usable across components
- Refactor in AudioSettings to keep gUM #uses stable.
* TODO: need to pass streams through AudioManager to avoid the surplus gUM.
- fix(audio): drop ScriptProcessorNode usage (deprecated)
* Used in volume meter for tracking - use hark instead
When setting skipCheckOnJoin to true, an extra audio channel is created in
FreeSWITCH, after user accepts the echo test. The extra channel is removed
when user leaves the room, but this still may affect performance.
When joining/returning breakouts, audio would always connect
with full audio. This can lead to a performance problem, once
all listenonly users would join full audio, increasing the
number of streams in FreeSWITCH.
We now have a consistent behavior, which is:
1 - The choice made by the user in the main room is predominant:
if mic is active in main room, user will automatically
join mic in breakout room. When returning from breakout
room, user will also join with mic again.
2 - Changes made in breakout room won't have effect when
returning to the main room. This means if user, for example,
change from listenonly to mic in breakout room, the returning
will consider the option choosen previously (listenonly) and
listenonly will be active again in the main room.
3 - If user didn't join audio in the main room, the audio modal
will be prompted when joining the breakout room (this is
a special case of (1))
The following is some technicall information:
InputStreamLiveSelector (component.jsx) now calls
'handleLeaveAudio' function, which is the default
function when user leaves audio (also used when
dynamic devices are inactive).
We now store information about user's choice (mic or listenonly)
using local storage, instead of the previous cookie method (this
was triggering some warnings in browser's console).
Also did a small refactoring to match eslint rules.
Fixes#11662.
